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Overview

RTMP (Real-Time Messaging Protocol) is a widely-supported protocol for live streaming. It’s the de facto standard for streaming from encoders like OBS, vMix, and Wirecast to media servers.

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Key Benefits

  • Wide compatibility: Supported by virtually all streaming software
  • Stable: Mature protocol with proven reliability
  • Simple setup: Easy to configure in most encoders
  • Low CPU usage: Efficient encoding and transmission
  • Codec support: H.264 video, AAC/MP3 audio

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Publishing Streams

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OBS Studio

Settings → Stream:
  Service: Custom
  Server: rtmp://your-server.com/WebRTCAppEE
  Stream Key: stream123

Settings → Output:
  Output Mode: Advanced
  Encoder: x264
  Rate Control: CBR
  Bitrate: 2500 Kbps
  Keyframe Interval: 2
  Preset: veryfast
  Profile: main
  Tune: zerolatency

Settings → Video:
  Base Resolution: 1920x1080
  Output Resolution: 1280x720
  FPS: 30

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Mobile Apps

import HaishinKit
import AVFoundation

class RTMPPublisher {
    let rtmpConnection = RTMPConnection()
    let rtmpStream: RTMPStream
    
    init() {
        rtmpStream = RTMPStream(connection: rtmpConnection)
        
        // Video settings
        rtmpStream.videoSettings = [
            .width: 1280,
            .height: 720,
            .bitrate: 2500 * 1000,
            .profileLevel: kVTProfileLevel_H264_Main_AutoLevel,
            .maxKeyFrameIntervalDuration: 2
        ]
        
        // Audio settings
        rtmpStream.audioSettings = [
            .bitrate: 128 * 1000,
            .sampleRate: 44100
        ]
    }
    
    func startPublishing() {
        rtmpStream.attachAudio(AVCaptureDevice.default(for: .audio))
        rtmpStream.attachCamera(DeviceUtil.device(withPosition: .back))
        
        rtmpConnection.connect("rtmp://your-server.com/WebRTCAppEE")
        rtmpStream.publish("stream123")
    }
    
    func stopPublishing() {
        rtmpStream.close()
        rtmpConnection.close()
    }
}

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Playing Streams

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Direct RTMP Playback

# Play RTMP stream
ffplay rtmp://your-server.com/WebRTCAppEE/stream123

# Play with low latency
ffplay -fflags nobuffer -flags low_delay \
  rtmp://your-server.com/WebRTCAppEE/stream123
RTMP playback in browsers requires Flash, which is deprecated. For browser playback, convert RTMP to HLS, DASH, or WebRTC using Ant Media Server’s transcoding capabilities.

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Configuration Options

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Server Configuration

# RTMP settings in application.properties

# Enable/disable RTMP
rtmp.enabled=true

# RTMP port (default: 1935)
rtmp.port=1935

# RTMP ingestion buffer time (ms)
settings.rtmpIngestBufferTimeMs=0

# Accept undefined streams
settings.acceptOnlyStreamsInDataStore=false

# Recording settings
settings.mp4MuxingEnabled=true
settings.hlsMuxingEnabled=true

# Adaptive bitrate
settings.adaptiveResolutionsEnabled=true

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Encoder Settings Recommendations

For Low Latency (3-5 seconds): 
Keyframe Interval: 1-2 seconds
Bitrate Mode: CBR
Profile: baseline or main
Tune: zerolatency
Buffer Size: 2x bitrate
For High Quality: 
Keyframe Interval: 2-4 seconds  
Bitrate Mode: VBR or CBR
Profile: high
Preset: medium or slow
Two-Pass Encoding: enabled
For Low Bandwidth: 
Resolution: 854x480 or lower
Bitrate: 500-1000 Kbps
Keyframe Interval: 2 seconds
Preset: veryfast or ultrafast
Profile: baseline

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Stream Authentication

// Enable token-based security
settings.acceptOnlyStreamsInDataStore=true

// Publish URL with token
rtmp://your-server.com/WebRTCAppEE/stream123?token=YOUR_TOKEN
Generate publish tokens via REST API: 
curl -X PUT "https://your-server.com:5443/WebRTCAppEE/rest/v2/broadcasts/stream123" \
  -H "Content-Type: application/json" \
  -d '{
    "name": "My Stream",
    "type": "liveStream",
    "streamId": "stream123"
  }'

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Troubleshooting

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Connection Issues

Cannot connect to RTMP server: 
# Test RTMP port
telnet your-server.com 1935

# Check firewall
sudo ufw status
sudo ufw allow 1935/tcp

# Verify Ant Media Server is running
sudo systemctl status antmedia
Stream not appearing: 
# List active streams
curl http://your-server.com:5080/WebRTCAppEE/rest/v2/broadcasts/list/0/10

# Check server logs
tail -f /usr/local/antmedia/log/antmedia-error.log

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Quality Issues

Pixelation/artifacts:
  • Increase bitrate
  • Use slower encoding preset
  • Check network bandwidth
  • Reduce resolution
Buffering/stuttering:
  • Decrease bitrate
  • Use faster preset (ultrafast/veryfast)
  • Check upload bandwidth
  • Reduce keyframe interval
High latency:
  • Set keyframe interval to 1-2 seconds
  • Enable tune=zerolatency in encoder
  • Use CBR bitrate mode
  • Reduce server-side buffer: rtmpIngestBufferTimeMs=0

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Debug Commands

# Monitor RTMP connections
netstat -an | grep 1935

# Check stream info with FFprobe
ffprobe -v quiet -print_format json -show_streams \
  rtmp://your-server.com/WebRTCAppEE/stream123

# Test stream with FFmpeg
ffmpeg -v verbose -i rtmp://your-server.com/WebRTCAppEE/stream123 \
  -t 10 -f null -

# Monitor bandwidth
iftop -i eth0

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Best Practices

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Production Recommendations

  1. Use consistent keyframe intervals: Set to 2 seconds for optimal HLS/DASH compatibility
  2. Enable CBR for live streaming: Provides predictable bandwidth usage
  3. Set appropriate buffer sizes: bufsize = 2 * bitrate
  4. Use authentication: Enable token-based security for production
  5. Monitor stream health: Use webhooks or REST API for monitoring
  6. Configure reconnection: Set up automatic reconnection in encoders

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Quality Presets

ffmpeg -i input.mp4 \
  -c:v libx264 -preset medium -b:v 4500k -maxrate 4500k -bufsize 9000k \
  -vf scale=1920:1080 -g 60 -keyint_min 60 \
  -c:a aac -b:a 192k -ar 48000 \
  -f flv rtmp://your-server.com/WebRTCAppEE/stream123

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Converting RTMP to Other Formats

Ant Media Server automatically converts RTMP to HLS, DASH, and WebRTC. Enable in configuration: 
settings.hlsMuxingEnabled=true
settings.dashMuxingEnabled=true
settings.webRTCEnabled=true
Access converted streams:
  • HLS: https://your-server.com:5443/WebRTCAppEE/streams/stream123.m3u8
  • DASH: https://your-server.com:5443/WebRTCAppEE/streams/stream123.mpd
  • WebRTC: Use WebRTC player with streamId: stream123

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